US9100467B2 - Alternate routing of voice calls in a heavily loaded SIP network - Google Patents
Alternate routing of voice calls in a heavily loaded SIP network Download PDFInfo
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- US9100467B2 US9100467B2 US13/419,262 US201213419262A US9100467B2 US 9100467 B2 US9100467 B2 US 9100467B2 US 201213419262 A US201213419262 A US 201213419262A US 9100467 B2 US9100467 B2 US 9100467B2
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- 238000004891 communication Methods 0.000 claims abstract description 82
- 238000000034 method Methods 0.000 claims abstract description 24
- 230000000977 initiatory effect Effects 0.000 claims abstract description 17
- 230000004044 response Effects 0.000 claims description 4
- 230000011664 signaling Effects 0.000 description 16
- 238000013500 data storage Methods 0.000 description 8
- 230000006870 function Effects 0.000 description 8
- 230000001960 triggered effect Effects 0.000 description 4
- 230000008859 change Effects 0.000 description 3
- 230000008901 benefit Effects 0.000 description 2
- 238000012986 modification Methods 0.000 description 2
- 230000004048 modification Effects 0.000 description 2
- 230000008569 process Effects 0.000 description 2
- 239000007787 solid Substances 0.000 description 2
- 230000002411 adverse Effects 0.000 description 1
- 238000013459 approach Methods 0.000 description 1
- 230000005540 biological transmission Effects 0.000 description 1
- 238000006243 chemical reaction Methods 0.000 description 1
- 230000003247 decreasing effect Effects 0.000 description 1
- 230000009977 dual effect Effects 0.000 description 1
- 230000000694 effects Effects 0.000 description 1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
- H04M7/0081—Network operation, administration, maintenance, or provisioning
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- H04L65/1006—
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/102—Gateways
- H04L65/1023—Media gateways
- H04L65/103—Media gateways in the network
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1069—Session establishment or de-establishment
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1104—Session initiation protocol [SIP]
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
Definitions
- the invention relates to the alternate routing of calls in a SIP network.
- IP networks generally provide an excellent infrastructure for geographically distributing components of a telecommunication system.
- the underlying IP network is optimal for transmission for control signaling, and when bandwidth is available, can provide an acceptable quality of service (or QoS) or grade of service (or GoS) for voice communications.
- QoS quality of service
- GoS grade of service
- a system had multiple communication gateways controlled by a single controller and the private switching facilities inter-connecting these gateways failed, users can “dial-out” on a public network trunk using the public address (or Direct Inward Dialing or DID number) of the destination party.
- This approach requires manual intervention by the user first to recognize that a problem exists, second to determine how to circumvent it, and third to dial the DID number. Normally, the calling party would dial only an extension to reach the destination party. If the destination party to be reached does not have a public number, he or she is not reachable by the alternate network.
- each enterprise may have a separate, independent, and active or primary media servers with resident call controller functionality, a plurality of digital stations, a plurality of IP or Internet Protocol stations, a gateway and a Local Area Network or LAN.
- the media servers are independent in that one media server in one enterprise is generally unaware of the subscriber configuration information, such as extensions, of the other enterprise's subscribers.
- the gateways are interconnected by the Public Switched Telephone Network or PSTN and Wide Area Network or WAN.
- the originating call controller determines the currently measured network delay and packet loss. When either measured variable reaches a predetermined threshold, the call controller automatically takes the idle IP trunk ports out-of-service, i.e., it busies out the ports. The ports remain out-of-service until the measurements return to the low threshold. No new calls are allowed over the IP trunk. Normal or conventional call routing over the PSTN is used for access to the next preference in the route pattern.
- the signaling channel for a call is routed over the WAN while the bearer channel is routed over the PSTN.
- the signaling channel in SBS includes SBS call-control signaling and QSIG private-networking protocol information.
- SBS associates the signaling and bearer channels at the SBS originating and terminating nodes so that they appear to the end users to be a normal, non-separated call.
- the use of the WAN for signaling traffic and the PSTN for voice bearer traffic addresses a customer need for using small amounts of bandwidth in the IP WAN for signaling traffic, with the voice bearer portion of the call being sent over inexpensive PSTN facilities.
- PSTN FallbackTM and SBSTM address architectures where there exist multiple, separate system implementations inter-connected by a traditional inter-switch trunking protocol; in other words, they permit inter-connection only of peer-to-peer systems.
- IP trunks and SBS With the move to larger, single-server IP WAN-connected media gateway distributed systems, there is no longer a need for IP trunks and SBS.
- Using trunk group administration to limit bandwidth between media servers is not required nor is PSTN FallbackTM when the number of calls exceeds the administered IP trunk member limit.
- Another technique for managing IP bandwidth usage includes call admission control in which the number of calls across the WAN or the bandwidth available for voice calls is limited. Call admission control can result in the call being denied and being forwarded to the callee's voice mail server (if accessible), thereby causing caller frustration.
- a communication manager providing a call admission control function can perform alternate routing, but not over the PSTN.
- alternate routing can be performed over the PSTN, but not on behalf of session initiation protocol (SIP) endpoints because such endpoints are not controlled by such communication servers.
- the communication server merely functions as a feature server for the SIP endpoints.
- inventions of the present disclosure enable the establishment of alternate routes for calls placed between endpoints, including but not limited to session initiation protocol (SIP) endpoints. More particularly, embodiments of the present disclosure allow for the routing of calls between endpoints using the public switched telephony network (PSTN). Moreover, at least some embodiments allow for calls to be completed in a reverse direction.
- PSTN public switched telephony network
- the present disclosure provides a solution that does not require tight control of the gateway by a communications server. For example, the creation of required state information on the gateways can be accomplished using standard SIP signaling.
- Embodiments of the present disclosure provide a communications core that controls and/or provides certain network communication features. More particularly, the communications core can comprise a server that implements one or more session managers or modules. In addition, the communications core can execute or provide an inter-gateway alternate route (IGAR) function or application. The session manager module or modules and the IGAR application can be executed by a communications core comprising a single server. In accordance with alternate embodiments, various components can be distributed among different server devices.
- the communications core is generally interconnected to a plurality of enterprise systems or network regions by a communication network.
- the communication network can comprise an Internet protocol network.
- the enterprise regions generally include a plurality of endpoints, such as SIP endpoints, and are associated with at least one terminal gateway.
- the different enterprise regions can be interconnected to one another by an alternate network, such as the PSTN, in addition to the IP communication network.
- Methods in accordance with embodiments of the present disclosure include invoking an IGAR application by a session manager module in response to a request to initiate a call, where the primary IP communication network is intact but overloaded and therefore unable to handle an additional call.
- the IGAR application in response to receiving a message from the session manager requesting initiation of a call, selects outgoing and incoming terminal gateways for the call, based on where they are relative to the call endpoints. The selection of the terminal gateways can also be made with reference to a cost index assigned to or otherwise associated with the terminal gateways.
- the IGAR application initiates the trunk call over the alternate network, such as the PSTN, using a unique identifier, such as a caller ID, that is assigned to a user in the calling branch, and using as the destination of the trunk call a unique direct inward dial (DID) number assigned to the destination branch.
- a unique identifier such as a caller ID
- DID direct inward dial
- the unique identifier and/or the unique DID number are not required to be dedicated numbers.
- the IGAR application can select any DID from the called and/or calling branch.
- the two legs or trunk calls can be matched based on caller ID.
- this solution is faster and potentially cheaper than previous solutions that utilized DTMF signaling to match different legs of a call.
- the connectivity information for the outgoing trunk is sent to the initiating or calling SIP endpoint, and ringback is provided to the user of the calling SIP endpoint.
- that terminal gateway notifies the session manager about the incoming trunk call.
- the session manager receiving that notification in turn notifies the IGAR application.
- the IGAR application uses the incoming identifier, the IGAR application matches the outgoing trunk to the incoming trunk.
- the IGAR application rings the called SIP endpoint and sends that endpoint the connectivity information needed to connect to the incoming call when the user of the called SIP endpoint answers.
- the user of the called SIP endpoint answers the user of the calling SIP endpoint is already connected to the trunk, and can hear the greeting from the user of the called SIP endpoint.
- the IGAR application directs the calling SIP endpoint to play local ringback.
- the IGAR application receives notification from the session manager and uses the incoming identifier to match the outgoing trunk to the incoming trunk.
- the IGAR application then sends the calling endpoint the connectivity information needed to connect to the outgoing trunk when the user of the called SIP endpoint answers.
- the IGAR application sends the connectivity information for the incoming trunk to the called SIP endpoint, and the calling user can hear the greeting from the user of the called SIP endpoint.
- FIG. 1 illustrates components of a communication network in accordance with embodiments of the present disclosure
- FIG. 2 illustrates components of an exemplary communication core server in accordance with embodiments of the present disclosure
- FIG. 3 illustrates aspects of a method for providing alternate call routing in a communication system in accordance with embodiments of the present disclosure.
- FIG. 1 illustrates an exemplary communication system 100 .
- the system 100 includes a communication server 104 and a plurality of network branches or regions 108 .
- the communication server 104 and the different network branches are interconnected to one another by an Internet protocol (IP) communication network 112 .
- IP Internet protocol
- the IP communication network 112 can comprise a plurality of networks.
- the IP communication network 112 can comprise the public Internet, a corporate wide area network (WAN), and/or a local area network (LAN).
- the branches 108 of the communication system 100 can comprise different enterprise regions or federated enterprises.
- the network branches 108 are interconnected by the public switched telephony network (PSTN) 116 .
- PSTN public switched telephony network
- the communication server 104 generally includes one or more session manager applications or modules 120 .
- a session manager application 120 can operate to control aspects of the operation of endpoints 132 , including but not limited to session initiation protocol endpoints, associated with a network branch 108 , and to provide various communication features to such endpoints.
- the communication server 104 can include an inter-gateway alternate routing (IGAR) application or module 124 .
- IGAR inter-gateway alternate routing
- the IGAR application 124 operates in cooperation with the one or more session managers 120 to provide alternate routing functionality over the PSTN 116 .
- the various modules, such as the session managers 120 and IGAR application 124 are typically distributed among different communication server 104 devices that are or can be in communication with one another.
- Each network region or branch 108 can include a terminal gateway 128 that operates to interconnect the associated branch 108 to the IP communication network 112 and/or the PSTN 116 .
- a network branch 108 can be associated with multiple trunk or terminal gateways 128 .
- a first terminal gateway 128 of a branch 108 can provide a connection to the IP communication network 112
- a second terminal gateway 128 of the region or branch 108 can provide connectivity to the PSTN 116 .
- each network branch 108 can include a plurality of endpoints or devices 132 , such as session initiation protocol (SIP) communication endpoints or devices.
- SIP session initiation protocol
- FIG. 2 illustrates components of an exemplary communication server 104 .
- a communication server 104 can comprise a general purpose computer or server device that provides various communication functions and features.
- the communication server 104 can include a processor 204 .
- the processor 204 may comprise a general purpose programmable processor or controller for executing application programming or instructions.
- the processor 204 may comprise a specially configured application specific integrated circuit (ASIC) or other integrated circuit, a digital signal processor, a hard wired electronic or logic circuit such as a discrete element circuit, a programmable logic device or gate array, special purpose computer, or the like.
- ASIC application specific integrated circuit
- the processor 204 generally functions to run programming code or instructions implementing various functions of the communication server.
- a communication server 104 may also include memory 208 for use in connection with the execution of application programming by the processor 204 , and for the temporary or long term storage of program instructions and/or data.
- the memory 208 may comprise RAM, SDRAM or other solid state memory.
- data storage 212 may be provided.
- data storage 212 can contain programming code or instructions implementing various of the applications or functions executed or performed by the communication server 104 .
- the data storage 212 may comprise a solid state memory device or devices.
- the data storage 212 may comprise a hard disk drive or other random access memory.
- the data storage 212 can include various applications and data.
- the data storage 212 can include a session manager application 220 , the execution of which by the processor 204 implements functions of a session manager module 120 .
- the session manager application 220 provides connectivity and features with respect to endpoints 132 associated with federated or other network branches 108 .
- the session manager application 220 can implement or provide a call admission control (CAC) application 224 .
- CAC call admission control
- the data storage 212 can include an IGAR application 124 .
- execution of the IGAR application 124 by the processor 204 enables communications between endpoints 132 in different network regions 108 to be completed over the PSTN 116 , where the primary IP communication network 112 is intact but unavailable to handle an additional call, such as where the primary IP communication network 112 is determined to be busy.
- the data storage 212 can also include cost index data 228 .
- the communication server 104 can additionally include a communication interface or interfaces 232 .
- the communication interface 232 may provide connectivity between the communication server 104 and other components of the network system 100 , including components of the different network branches 108 , via the IP communication network or networks 112 .
- the session manager 120 responsible for the network region 108 from which the call originated begins routing the call (step 308 ).
- the routing of the call can include determining by the CAC application 224 for or associated with the session manager application 120 whether the call can be completed normally, over the IP communication network or networks 112 . If the CAC application 224 indicates that call limits have been exceeded or that alternate routing is otherwise needed, the IGAR application 124 is triggered.
- the invite message When the alternate call is initiated in the reverse direction, the invite message includes the unique identifier of or assigned to the called party and a unique DID number assigned to the calling branch 108 .
- the selection of terminal gateways can include determining where the trunks are relative to the endpoints 132 involved in the call.
- the trunks can be traditional time division multiplexing (TDM) trunks or may represent connections over other alternate networks.
- TDM time division multiplexing
- the session manager 120 launches the call via the selected, first terminal gateway 128 .
- the IGAR application 124 can initiate the trunk call by sending an invite message to the session manager 120 associated with the terminal gateway 128 that has been selected by the IGAR application 124 to launch the trunk call. That is, the trunk call can be originated from the session manager 120 and the terminal gateway 128 of either the calling or the called branch 108 .
- the selection of a branch 108 from which to launch the trunk call can be made by selecting the lower cost branch 108 for originating the trunk call, for example by referencing a cost index (cost index data 228 ) stored on or available to the communication server 104 .
- the selected session manager initiates the trunk call it uses a unique caller identifier (ID) assigned by the IGAR application 124 .
- ID unique caller identifier
- the call routes across the PSTN 116 to the second terminal gateway 128 . More particularly, the session manager 120 removes the phase tag from the invite message, which identifies the communication endpoint 132 and the listed directory number (LDN) of the IGAR application 124 , and launches the call via the first terminal gateway 128 to the second terminal gateway 128 .
- the second terminal gateway 128 notifies the session manager 120 of the incoming call (step 336 ).
- the session manager 120 associated with the second terminal gateway 128 recognizes the LDN of the IGAR application 124 , and delivers the call to the IGAR application 124 (step 344 ).
- the IGAR application 124 uses the incoming identifier or caller ID to match the outgoing trunk to the incoming trunk (identified by the unique DID number), and sends an invite message to the called endpoint 132 with the connectivity information needed by the called endpoint 132 to connect that endpoint to the incoming trunk at the second terminal gateway 128 (step 348 ).
- This connectivity information can include the incoming trunk session description protocol (SDP) information.
- SDP trunk session description protocol
- the user associated with the called endpoint 132 can then pick up and be placed in communication with the user associated with the calling endpoint 132 over the alternate route. Accordingly, information required to match the incoming and outgoing trunks is communicated as part of the signaling used in establishing the alternate call routing.
- embodiments of the present disclosure allow alternate call routing to be performed using SIP endpoints, or non-SIP endpoints that are serviced by SIP-enabled intermediary nodes or devices. Calls can also be routed using any terminal gateway 128 within the enterprise network of the associated communication endpoint 132 , including terminal gateways 128 in neighboring branches 108 .
- Embodiments of the present disclosure also provide support for launching alternate call routes in the “reverse” direction.
- the IGAR application 124 can cause an alternate call to be launched by the terminal gateway 128 at the called endpoint 132 end. Whether the terminal gateway 128 at the calling or called end launches the alternate call can be specified by a system administrator. Alternatively, a least expensive direction can be selected by the IGAR application 124 by referencing the cost index data 228 .
- systems and methods disclosed herein allow for the ultimate routing of a session initiation protocol call over the PSTN 116 using any terminal gateway 128 , including a SIP capable terminal gateway 128 , in a communication system 100 , including terminal gateways 128 and neighboring or otherwise interconnected branches 108 .
- the use of unique identifiers passed via trunk signaling allows incoming and outgoing trunks to be paired up by the IGAR application 124 . Moreover, this is accomplished without requiring additional out of band signaling or in band DTMF signaling.
- the direction of the trunk call can be controlled, allowing a lower cost trunk to be selected as the originating, even where the lower cost trunk is associated with the called endpoint 132 network branch 108 .
- endpoints 132 comprising SIP endpoints or devices
- embodiments of the disclosed invention are not so limited.
- embodiments can include any system in which one or both endpoints (e.g., phones) 132 are not SIP endpoints and are connected to a session manager 120 through some gateway 128 or access element that provides conversion between non-SIP protocol from/to the endpoint 132 and SIP to/from the session manager 120 .
- an endpoint 132 can include any endpoint that communicates using SIP either directly or via an intermediary device.
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Abstract
Description
-
- a) SIP user A (endpoint A 132) calls SIP user B (endpoint B 132)
- b) Session Manager (SM) 120 uses Call Admission Control (CAC) 224 to determine that there is not enough bandwidth to complete the call. It does not ring SIP user B at this point.
- c)
SM 120 initiates alternate routing by sending a message to anintegrated IGAR application 124. - d) The
IGAR application 124 selects the outgoing and incoming terminal gateways (TGs) 128 for the call, based on where they are relative to the twoendpoints 132, and a “cost index” assigned to eachTG 128.- a. These trunks may be traditional TDM trunks, or may represent connections over an alternate data network.
- b. The two TGs may be in the same branch as each of the parties, or may be in a neighboring branch.
- e) The
IGAR application 124 initiates the trunk call, using a unique caller ID assigned to one of the users in the calling branch. The destination of the trunk call is a unique DID number assigned to the destination branch. - f) The
IGAR application 124 sends the connectivity information for the outgoing trunk to SIP endpoint A, and the user begins to hear ringback from the trunk. - g) When the trunk call arrives at the
TG 132 in the other branch, theTG 132 notifiesSM 120 about the incoming trunk call, which in turn notifies theIGAR application 124. - h) The
IGAR application 124 uses the incoming caller ID to match the outgoing trunk to the incoming trunk. - i) The
IGAR application 124 ringsSIP endpoint B 132, sending it the connectivity information it needs to connect to the incoming trunk when SIP user B answers. - j) When SIP user B answers, SIP user A is already connected to the trunk and can hear the “hello” greeting from SIP user B.
-
- f) The
IGAR application 124 directs SIP user A′s endpoint to play local ringback. - g) No change from above
- h) No change from above
- i) The
IGAR application 124 ringsSIP endpoint B 132, sending it the connectivity information it needs to connect to the outgoing trunk when SIP user B answers. - j) When SIP user B answers, the
IGAR application 124 sends the connectivity information for the incoming trunk toSIP endpoint A 132, and the user can hear the “hello” greeting from SIP user B.
- f) The
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US13/419,262 US9100467B2 (en) | 2011-12-27 | 2012-03-13 | Alternate routing of voice calls in a heavily loaded SIP network |
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US201161580554P | 2011-12-27 | 2011-12-27 | |
US13/419,262 US9100467B2 (en) | 2011-12-27 | 2012-03-13 | Alternate routing of voice calls in a heavily loaded SIP network |
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US20130163433A1 US20130163433A1 (en) | 2013-06-27 |
US9100467B2 true US9100467B2 (en) | 2015-08-04 |
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US9426021B2 (en) | 2013-04-29 | 2016-08-23 | Avaya Inc. | Communication failover in a distributed network |
US20160036985A1 (en) * | 2014-07-30 | 2016-02-04 | Nir Koren | Real-time rule-based recovery platform |
US9735981B2 (en) | 2015-03-03 | 2017-08-15 | At&T Intellectual Property I, L.P. | Facilitation of session initiation protocol trunking |
GB2583702B (en) * | 2019-04-18 | 2021-05-12 | Metaswitch Networks Ltd | Call control in telephony-capable communications networks |
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US20130163433A1 (en) | 2013-06-27 |
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